On 16/05/2024 12:26, Andreas Rheinhardt wrote:
Lynne via ffmpeg-devel:This commit adds a decoder for the frequency-domain part of USAC.What works: - Mono - Stereo (no prediction) - Stereo (mid/side coding) What doesn't: - Preroll decoding (every single decoder seems faulty or weird?) - Complex stereo prediction Known issues: - Spec incompliance (noise synthesis in particular) - Lack of robustness ---diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index 20545a24d4..3e6592cf0e 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -42,6 +42,8 @@ #include "libavcodec/avcodec.h" #include "libavcodec/mpeg4audio.h"+#include "aacdec_ac.h"+ typedef struct AACDecContext AACDecContext;/**@@ -69,6 +71,32 @@ enum CouplingPoint { AFTER_IMDCT = 3, };+enum AACUsacElem {+ ID_USAC_SCE = 0, + ID_USAC_CPE = 1, + ID_USAC_LFE = 2, + ID_USAC_EXT = 3, +}; + +enum ExtensionHeaderType { + ID_CONFIG_EXT_FILL = 0, + ID_CONFIG_EXT_LOUDNESS_INFO = 2, + ID_CONFIG_EXT_STREAM_ID = 7, +}; + +enum AACUsacExtension { + ID_EXT_ELE_FILL, + ID_EXT_ELE_MPEGS, + ID_EXT_ELE_SAOC, + ID_EXT_ELE_AUDIOPREROLL, + ID_EXT_ELE_UNI_DRC, +}; + +enum AACUSACLoudnessExt { + UNIDRCLOUDEXT_TERM = 0x0, + UNIDRCLOUDEXT_EQ = 0x1, +}; + // Supposed to be equal to AAC_RENAME() in case of USE_FIXED. #define RENAME_FIXED(name) name ## _fixed@@ -93,6 +121,40 @@ typedef struct LongTermPrediction {int8_t used[MAX_LTP_LONG_SFB]; } LongTermPrediction;+/* Per channel core mode */+typedef struct AACUsacElemData { + uint8_t core_mode; + uint8_t scale_factor_grouping; + + /* Timewarping ratio */ +#define NUM_TW_NODES 16 + uint8_t tw_ratio[NUM_TW_NODES]; + + struct { + uint8_t acelp_core_mode : 3; + uint8_t lpd_mode : 5; + + uint8_t bpf_control_info : 1; + uint8_t core_mode_last : 1; + uint8_t fac_data_present : 1; + + int last_lpd_mode; + } ldp; + + struct { + unsigned int seed; + uint8_t level : 3; + uint8_t offset : 5; + } noise; + + struct { + uint8_t gain; + uint32_t kv[8 /* (1024 / 16) / 8 */][8]; + } fac; + + AACArithState ac; +} AACUsacElemData; + /** * Individual Channel Stream */ @@ -145,6 +207,7 @@ typedef struct ChannelCoupling { */ typedef struct SingleChannelElement { IndividualChannelStream ics; + AACUsacElemData ue; ///< USAC element data TemporalNoiseShaping tns; enum BandType band_type[128]; ///< band types int sfo[128]; ///< scalefactor offsets @@ -163,25 +226,141 @@ typedef struct SingleChannelElement { }; } SingleChannelElement;+typedef struct AACUsacStereo {+ uint8_t common_window; + uint8_t common_tw; + + uint8_t ms_mask_mode; + uint8_t config_idx; + + struct { + uint8_t use_prev_frame; + uint8_t pred_dir; + uint8_t delta_code_time; + uint8_t pred_used[8][64]; + + AVComplexFloat pred[8][64]; + } cplx; +} AACUsacStereo; + /** * channel element - generic struct for SCE/CPE/CCE/LFE */ typedef struct ChannelElement { int present; // CPE specific + uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band // shared SingleChannelElement ch[2]; // CCE specific ChannelCoupling coup; + // USAC stereo coupling data + AACUsacStereo us; } ChannelElement;+typedef struct AACUSACLoudnessInfo {+ uint8_t drc_set_id : 6; + uint8_t downmix_id : 7; + struct { + uint16_t lvl : 12; + uint8_t present : 1; + } sample_peak; + + struct { + uint16_t lvl : 12; + uint8_t measurement : 4; + uint8_t reliability : 2; + uint8_t present : 1; + } true_peak; + + uint8_t nb_measurements : 4; + struct { + uint8_t method_def : 4; + uint8_t method_val; + uint8_t measurement : 4; + uint8_t reliability : 2; + } measurements[16]; +} AACUSACLoudnessInfo; + +typedef struct AACUsacElemConfig { + enum AACUsacElem type; + + uint8_t tw_mdct : 1; + uint8_t noise_fill : 1; + + uint8_t stereo_config_index; + + struct { + int ratio; + + uint8_t harmonic_sbr : 1; /* harmonicSBR */ + uint8_t bs_intertes : 1; /* bs_interTes */ + uint8_t bs_pvc : 1; /* bs_pvc */ + + struct { + uint8_t start_freq; /* dflt_start_freq */ + uint8_t stop_freq; /* dflt_stop_freq */ + + uint8_t freq_scale; /* dflt_freq_scale */ + uint8_t alter_scale : 1; /* dflt_alter_scale */ + uint8_t noise_scale; /* dflt_noise_scale */ + + uint8_t limiter_bands; /* dflt_limiter_bands */ + uint8_t limiter_gains; /* dflt_limiter_gains */ + uint8_t interpol_freq : 1; /* dflt_interpol_freq */ + uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */ + } dflt; + } sbr; + + struct { + uint8_t freq_res; /* bsFreqRes */ + uint8_t fixed_gain; /* bsFixedGainDMX */ + uint8_t temp_shape_config; /* bsTempShapeConfig */ + uint8_t decorr_config; /* bsDecorrConfig */ + uint8_t high_rate_mode : 1; /* bsHighRateMode */ + uint8_t phase_coding : 1; /* bsPhaseCoding */ + + uint8_t otts_bands_phase; /* bsOttBandsPhase */ + uint8_t residual_coding; /* bsResidualCoding */ + uint8_t residual_bands; /* bsResidualBands */ + uint8_t pseudo_lr : 1; /* bsPseudoLr */ + uint8_t env_quant_mode : 1; /* bsEnvQuantMode */Is using bitfields really worth it given that they force to use masking for accesses?+ } mps; + + struct { + enum AACUsacExtension type; + uint8_t payload_frag; + uint32_t default_len; + uint32_t pl_data_offset; + uint8_t *pl_data; + } ext; +} AACUsacElemConfig; + +typedef struct AACUSACConfig { + uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */ + uint8_t rate_idx; + uint16_t core_frame_len; + uint16_t stream_identifier; + + AACUsacElemConfig elems[64]; + int nb_elems; + + struct { + uint8_t nb_album; + AACUSACLoudnessInfo album_info[64]; + uint8_t nb_info; + AACUSACLoudnessInfo info[64]; + } loudness; +} AACUSACConfig; + typedef struct OutputConfiguration { MPEG4AudioConfig m4ac; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; AVChannelLayout ch_layout; enum OCStatus status; + AACUSACConfig usac; } OutputConfiguration;/**diff --git a/libavcodec/aac/aacdec_ac.c b/libavcodec/aac/aacdec_ac.c new file mode 100644 index 0000000000..326d716bd3 --- /dev/null +++ b/libavcodec/aac/aacdec_ac.c @@ -0,0 +1,224 @@ +/* + * AAC definitions and structures + * Copyright (c) 2024 Lynne + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavcodec/aactab.h" +#include "aacdec_ac.h" + +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int N) +{ + float ratio; + if (reset) { + memset(state->last, 0, sizeof(state->last)); + state->last_len = N; + memset(state->cur, 0, sizeof(state->cur)); + state->cur[3] = 0; + state->cur[2] = 0; + state->cur[1] = 0; + state->cur[0] = 1; + state->last[0] = 0 << 12; + state->state_pre = 0; + return 0; + } else if (state->last_len != N) { + int i; + uint8_t last[512 /* 2048 / 4 */]; + memcpy(last, state->last, sizeof(last)); + + ratio = state->last_len / (float)N; + for (i = 0; i < N/2; i++) { + int k = (int)(i * ratio); + state->last[i] = last[k]; + } + + for (; i < FF_ARRAY_ELEMS(state->last); i++) + state->last[i] = 0; + + state->last_len = N; + } + + memset(state->cur, 0, sizeof(state->cur)); + state->cur[3] = 0; + state->cur[2] = 0; + state->cur[1] = 0; + state->cur[0] = 1; + + state->state_pre = state->last[0] << 12; + return state->last[0] << 12; +} + +extern int ec_debug; + +int trig = 0; + +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t c, int i, int N) +{ + c = state->state_pre >> 8; + c = c + (state->last[i + 1] << 8); + c = (c << 4); + c += state->cur[1]; + + state->state_pre = c; + + if (i > 3 && + ((state->cur[3] + state->cur[2] + state->cur[1]) < 5)) + return c + 0x10000; + + return c; +} + +uint32_t ff_aac_ac_get_pk(uint32_t c) +{ + int i_min = -1; + int i, j; + int i_max = FF_ARRAY_ELEMS(ff_aac_ac_lookup_m) - 1; + while ((i_max - i_min) > 1) { + i = i_min + ((i_max - i_min) / 2); + j = ff_aac_ac_hash_m[i]; + if (c < (j >> 8)) + i_max = i; + else if (c > (j >> 8)) + i_min = i; + else + return (j & 0xFF); + } + return ff_aac_ac_lookup_m[i_max]; +} + +void ff_aac_ac_update_context(AACArithState *state, int idx, + uint16_t a, uint16_t b) +{ + state->cur[0] = a + b + 1; + if (state->cur[0] > 0xF) + state->cur[0] = 0xF; + + state->cur[3] = state->cur[2]; + state->cur[2] = state->cur[1]; + state->cur[1] = state->cur[0]; + + state->last[idx] = state->cur[0]; +} + +/* Initialize AC */ +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb) +{ + ac->low = 0; + ac->high = UINT16_MAX; + ac->val = get_bits(gb, 16); +} + +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, + const uint16_t *cdf, uint16_t cdf_len) +{ + int val = ac->val; + int low = ac->low; + int high = ac->high; + + int rng = high - low + 1; + int c = ((((int)(val - low + 1)) << 14) - ((int)1)); + + /* Note: this could be done faster via heuristics, the total number of + * configurations is low */ + const uint16_t *p = cdf - 1; + const uint16_t *q; + + switch (cdf_len) { + case 2: + if ((p[1] * rng) > c) + p += 1; + break; + case 4: + if ((p[2] * rng) > c) + p += 2; + if ((p[1] * rng) > c) + p += 1; + break; + case 17: + /* First check if the current probability is even met at all */ + if ((p[1] * rng) <= c) + break; + p += 1; + for (int i = 8; i >= 1; i >>= 1) + if ((p[i] * rng) > c) + p += i; + break; + case 27: + const uint16_t *p_24 = p + 24; + + if ((p[16] * rng) > c) + p += 16; + if ((p[8] * rng) > c) + p += 8; + if (p != p_24) + if ((p[4] * rng) > c) + p += 4; + if ((p[2] * rng) > c) + p += 2; + + if (p != &p_24[2]) + if ((p[1] * rng) > c) + p += 1; + break; + default: + /* This should never happen */ + av_assert2(0); + } + + int sym = (int)((ptrdiff_t)(p - cdf)) + 1; + if (sym) + high = low + ((rng * cdf[sym - 1]) >> 14) - 1; + low += (rng * cdf[sym]) >> 14; + + /* This loop could be done faster */ + while (1) { + if (high < 32768) { + ; + } else if (low >= 32768) { + val -= 32768; + low -= 32768; + high -= 32768; + } else if (low >= 16384 && high < 49152) { + val -= 16384; + low -= 16384; + high -= 16384; + } else { + break; + } + low += low; + high += high + 1; + val = (val << 1) | get_bits1(gb); + }; + + ac->low = low; + ac->high = high; + ac->val = val; + + return sym; +} + +void ff_aac_ac_finish(AACArithState *state, int offset, int N) +{ + int i; + + for (i = offset; i < N/2; i++) + state->last[i] = 1; + + for (; i < FF_ARRAY_ELEMS(state->last); i++) + state->last[i] = 0; +} diff --git a/libavcodec/aac/aacdec_ac.h b/libavcodec/aac/aacdec_ac.h new file mode 100644 index 0000000000..ef96bed770 --- /dev/null +++ b/libavcodec/aac/aacdec_ac.h @@ -0,0 +1,54 @@ +/* + * AAC definitions and structures + * Copyright (c) 2024 Lynne + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AACDEC_AC_H +#define AVCODEC_AACDEC_AC_H + +#include "libavcodec/get_bits.h" + +typedef struct AACArithState { + uint8_t last[512 /* 2048 / 4 */]; + int last_len; + uint8_t cur[4]; + uint16_t state_pre; +} AACArithState; + +typedef struct AACArith { + uint16_t low; + uint16_t high; + uint16_t val; +} AACArith; + +#define FF_AAC_AC_ESCAPE 16 + +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int len); +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t old_c, int idx, int len); +uint32_t ff_aac_ac_get_pk(uint32_t c); + +void ff_aac_ac_update_context(AACArithState *state, int idx, uint16_t a, uint16_t b); +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb); + +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, + const uint16_t *cdf, uint16_t cdf_len); + +void ff_aac_ac_finish(AACArithState *state, int offset, int nb); + +#endif /* AVCODEC_AACDEC_AC_H */ diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c index 59a69d88f3..8d31af22f8 100644 --- a/libavcodec/aac/aacdec_dsp_template.c +++ b/libavcodec/aac/aacdec_dsp_template.c @@ -88,8 +88,8 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs); const uint16_t *offsets = ics->swb_offset; for (int g = 0; g < ics->num_window_groups; g++) { - for (int sfb = 0; sfb < ics->max_sfb; sfb++) { - const int idx = g*ics->max_sfb + sfb; + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + const int idx = g*cpe->max_sfb_ste + sfb; if (cpe->ms_mask[idx] && cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h index e40a2fe1a7..047c11e0fb 100644 --- a/libavcodec/aac/aacdec_latm.h +++ b/libavcodec/aac/aacdec_latm.h @@ -56,7 +56,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, { AACDecContext *ac = &latmctx->aac_ctx; AVCodecContext *avctx = ac->avctx; - MPEG4AudioConfig m4ac = { 0 }; + OutputConfiguration oc = { 0 }; + MPEG4AudioConfig *m4ac = &oc.m4ac; GetBitContext gbc; int config_start_bit = get_bits_count(gb); int sync_extension = 0; @@ -76,7 +77,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, if (get_bits_left(gb) <= 0) return AVERROR_INVALIDDATA;- bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,+ bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc, &gbc, config_start_bit, sync_extension);@@ -88,11 +89,12 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,asclen = bits_consumed;if (!latmctx->initialized ||- ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || - ac->oc[1].m4ac.chan_config != m4ac.chan_config) { + ac->oc[1].m4ac.sample_rate != m4ac->sample_rate || + ac->oc[1].m4ac.chan_config != m4ac->chan_config) {if (latmctx->initialized) {- av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); + av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", + m4ac->sample_rate, m4ac->chan_config); } else { av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); } @@ -280,7 +282,7 @@ static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, } else { push_output_configuration(&latmctx->aac_ctx); if ((err = decode_audio_specific_config( - &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, + &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1], avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { pop_output_configuration(&latmctx->aac_ctx); return err; diff --git a/libavcodec/aac/aacdec_lpd.c b/libavcodec/aac/aacdec_lpd.c new file mode 100644 index 0000000000..be39e2c175 --- /dev/null +++ b/libavcodec/aac/aacdec_lpd.c @@ -0,0 +1,192 @@ +/* + * Copyright (c) 2024 Lynne <d...@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "aacdec_lpd.h" +#include "aacdec_usac.h" + +const uint8_t ff_aac_lpd_mode_tab[32][4] = { + { 0, 0, 0, 0 }, + { 1, 0, 0, 0 }, + { 0, 1, 0, 0 }, + { 1, 1, 0, 0 }, + { 0, 0, 1, 0 }, + { 1, 0, 1, 0 }, + { 0, 1, 1, 0 }, + { 1, 1, 1, 0 }, + { 0, 0, 0, 1 }, + { 1, 0, 0, 1 }, + { 0, 1, 0, 1 }, + { 1, 1, 0, 1 }, + { 0, 0, 1, 1 }, + { 1, 0, 1, 1 }, + { 0, 1, 1, 1 }, + { 1, 1, 1, 1 }, + { 2, 2, 0, 0 }, + { 2, 2, 1, 0 }, + { 2, 2, 0, 1 }, + { 2, 2, 1, 1 }, + { 0, 0, 2, 2 }, + { 1, 0, 2, 2 }, + { 0, 1, 2, 2 }, + { 1, 1, 2, 2 }, + { 2, 2, 2, 2 }, + { 3, 3, 3, 3 }, + /* Larger values are reserved, but permit them for resilience */ + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, +}; + +static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn) +{ + if (nk_mode == 1) { + for (int k = 0; k < no_qn; k++) { + qn[k] = ff_aac_get_vlclbf(gb); + if (qn[k]) + qn[k]++; + } + return; + } + + for (int k = 0; k < no_qn; k++) + qn[k] = get_bits(gb, 2) + 2; + + if (nk_mode == 2) { + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + qn[k] = ff_aac_get_vlclbf(gb); + if (qn[k]) + qn[k] += 4; + } + } + return; + } + + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + int qn_ext = ff_aac_get_vlclbf(gb); + switch (qn_ext) { + case 0: qn[k] = 5; break; + case 1: qn[k] = 6; break; + case 2: qn[k] = 0; break; + default: qn[k] = qn_ext + 4; break; + } + } + } +} + +static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv, + int nk_mode, int no_qn) +{ + int n, nk; + + int qn[2]; + parse_qn(gb, qn, nk_mode, no_qn); + + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + nk = (qn[k] - 3) / 2; + n = qn[k] - nk*2; + } else { + nk = 0; + n = qn[k]; + } + } + + int idx = get_bits(gb, 4*n); + + if (nk > 0) + for (int i = 0; i < 8; i++) + kv[i] = get_bits(gb, nk); + + return 0; +} + +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, + int use_gain, int len) +{ + int ret; + if (use_gain) + ce->fac.gain = get_bits(gb, 7); + + for (int i = 0; i < len/8; i++) { + ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1); + if (ret < 0) + return ret; + } + + return 0; +} + +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemData *ce, GetBitContext *gb) +{ + ce->ldp.acelp_core_mode = get_bits(gb, 3); + ce->ldp.lpd_mode = get_bits(gb, 5); + + ce->ldp.bpf_control_info = get_bits1(gb); + ce->ldp.core_mode_last = get_bits1(gb); + ce->ldp.fac_data_present = get_bits1(gb); + + const uint8_t *mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode]; + + int first_ldp_flag = !ce->ldp.core_mode_last; + int first_tcx_flag = 1; + if (first_ldp_flag) + ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */ + + int k = 0; + while (k < 0) { + if (!k) { + if (ce->ldp.core_mode_last && ce->ldp.fac_data_present) + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); + } else { + if (!ce->ldp.last_lpd_mode && mod[k] > 0 || + ce->ldp.last_lpd_mode && !mod[k]) + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); + } + if (!mod[k]) { +// parse_acelp_coding(); + ce->ldp.last_lpd_mode = 0; + k++; + } else { +// parse_tcx_coding(); + ce->ldp.last_lpd_mode = mod[k]; + k += (1 << (mod[k] - 1)); + first_tcx_flag = 0; + } + } + +// parse_lpc_data(first_lpd_flag); + + if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) { + uint16_t len_8 = usac->core_frame_len / 8; + uint16_t len_16 = usac->core_frame_len / 16; + uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16; + int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len); + if (ret < 0) + return ret; + } + + return 0; +} diff --git a/libavcodec/aac/aacdec_lpd.h b/libavcodec/aac/aacdec_lpd.h new file mode 100644 index 0000000000..924ff75e52 --- /dev/null +++ b/libavcodec/aac/aacdec_lpd.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2024 Lynne <d...@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AAC_AACDEC_LPD_H +#define AVCODEC_AAC_AACDEC_LPD_H + +#include "aacdec.h" +#include "libavcodec/get_bits.h" + +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, + int use_gain, int len); + +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemData *ce, GetBitContext *gb); + +#endif /* AVCODEC_AAC_AACDEC_LPD_H */ diff --git a/libavcodec/aac/aacdec_usac.c b/libavcodec/aac/aacdec_usac.c new file mode 100644 index 0000000000..4b48c4d6ca --- /dev/null +++ b/libavcodec/aac/aacdec_usac.c @@ -0,0 +1,1230 @@ +/* + * Copyright (c) 2024 Lynne <d...@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "aacdec_usac.h" +#include "aacdec_tab.h" +#include "aacdec_lpd.h" +#include "aacdec_ac.h" + +#include "libavcodec/opusdsp.h" +#include "libavcodec/aactab.h" +#include "libavutil/mem.h" +#include "libavcodec/mpeg4audio.h" + +/* Number of scalefactor bands per complex prediction band, equal to 2. */ +#define SFB_PER_PRED_BAND 2 + +static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, int nb3) +{ + uint32_t val = get_bits(gb, nb1); + if (val < ((1 << nb1) - 1)) + return val; + + val += get_bits(gb, nb2); + if (val == ((1 << nb2) - 1)) + val += get_bits(gb, nb3); + + return val; +} + +static int aac_usac_samplerate[] = {Missing const+ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, + 16000, 12000, 11025, 8000, 7350, -1, -1, 57600, 51200, + 40000, 38400, 34150, 28800, 25600, 20000, 19200, 17075, 14400, 12800, 9600, -1, -1, -1, -1, +}; ++static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e, + GetBitContext *gb) +{ + if (get_bits1(gb)) { /* usacExtElementPresent */ + uint32_t len; + if (get_bits1(gb)) { /* usacExtElementUseDefaultLength */ + len = e->ext.default_len; + } else { + len = get_bits(gb, 8); /* usacExtElementPayloadLength */ + if (len == 255) + len += get_bits(gb, 16) - 2; + } + + if (len) { + uint8_t *tmp; + uint8_t pl_frag_start = 1; + uint8_t pl_frag_end = 1; + if (e->ext.payload_frag) { + pl_frag_start = get_bits1(gb); /* usacExtElementStart */ + pl_frag_end = get_bits1(gb); /* usacExtElementStop */ + } + + if (pl_frag_start) + e->ext.pl_data_offset = 0; + + tmp = av_realloc(e->ext.pl_data, e->ext.pl_data_offset + len); + if (!tmp) { + free(e->ext.pl_data);Wrong deallocator.+ return AVERROR(ENOMEM); + } + e->ext.pl_data = tmp; + + for (int i = 0; i < len; i++) + e->ext.pl_data[e->ext.pl_data_offset + i] = get_bits(gb, 8); + + if (pl_frag_end) { + int ret; + e->ext.pl_data_offset = 0; + switch (e->ext.type) { + case ID_EXT_ELE_FILL: + av_freep(&e->ext.pl_data); + break; + case ID_EXT_ELE_AUDIOPREROLL: + ret = parse_audio_preroll(ac, e->ext.pl_data, + e->ext.pl_data_offset); + if (ret < 0) { + av_freep(&e->ext.pl_data); + return ret; + } + break; + default: + av_freep(&e->ext.pl_data);Pointless if you abort in the next line+ /* This should never happen */ + av_assert0(0); + } + } + } + } + + return 0; +} ++#include "libavcodec/opusdsp.h" + +#ifndef AVCODEC_AAC_AACDEC_USAC_H +#define AVCODEC_AAC_AACDEC_USAC_H + +#include "aacdec.h" + +#include "libavcodec/get_bits.h" + +static inline uint8_t ff_aac_get_vlclbf(GetBitContext *gb) +{ + uint8_t ret = 0; + while (get_bits1(gb) && ret <= 36) + ret++; + return ret; +}Look at unary.h
That's convenient, thanks. I've synced my changes on my repo in https://github.com/cyanreg/FFmpeg/tree/xheThough I'll likely upload the patchset on the ML again tomorrow with some fixes for preroll parsing and complex synth.
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